WARNING: this is an automatic post retrieved from the Asterisk-Users Mailing List, not an authored post
Mailing-list Collector May 27, 2012Asterisk Users7 Comments

Hi list,

we are upgrading our Asterisk production server from 1.6.24 to 1.8.12
version and face the following problem: one of our peer
(voicetrading.com) doesn’t accept our calls anymore, we receive a
timeout error “Packet timed out after 32000ms with no response”.

Switching back to 1.6 make things working again!

In sip.conf we have nat=no, peer conf is:

[myPeerDef]
type=peer
host=111.111.1.111
context=honeypot

insecure=invite

directmedia=no

disallow=all

allow=ulaw,alaw

dtmfmode=inband

We aren’t registered, our IP is authorized by their system.

Debug of sessions (222.222.22.22 is our server 111.111.1.111 is their)

WARNING: this is an automatic post retrieved from the Asterisk-Users Mailing List, not an authored post
Mailing-list Collector April 14, 2012Asterisk Users2 Comments

This is a really simple problem that I just can’t get to work. There
are two Asterisk servers with the following sip user and peer. When a
call is attempted, Asterisk is not sending authentication details in
response to the 401. Note, if the secret is blank on 172.16.0.2 test,
the INVITE succeeds.

The error message is misleading; you are having this problem because the
‘m’ line in the SDP with the ‘audio’ offer has a port number of 0
(zero)., which means it is not an active media stream offer. It does not
make any sense for the SDP in an INVITE for a new call to have an m-line
with a port number of zero.

I’ll improve the error message so that this sort of situation won’t be
as confusing in the future.